Apparatus and systems which transmit, process or store audio and/or video signals in digital form are becoming more widespread. In order to be able to effect a signal transmission between different apparatus of this type, a sampling rate converter is required if the apparatus have interfaces which use different sampling rates. In the digital art, the sampling rate or the scanning frequency with which an analogue or else a digital signal is sampled, is one of the most important properties of the system. It indicates how quickly the individual samples succeed each other and it is also referred to as the sampling frequency. In order to capture the full information content of an analogue signal, the sampling frequency is selected, in accordance with the practically proven Shannon sampling theorem, to be at least twice as high as the frequency being measured. Audio signals in the frequency range up to 20 kHz would thus require a sampling frequency of 40 kHz. However, different sampling frequencies were laid down for the individual recording media and methods of transmission. Thus, for example, the sampling frequency amounts to:
44.1 kHz for the compact disc, PA1 48 kHz for digital audio tape and PA1 32 kHz for satellite broadcasting.
A sampling frequency convertor is required when processing the signals of a system using a first sampling frequency, in a system which uses a second sampling frequency, due to the different sampling time points caused by the differing frequencies, in order to provide samples in the raster of the second sampling frequency which correspond as closely as possible to the analogue signal occurring at a particular time point. Basically, two ways are available for this purpose. The digital signal is either converted into an analogue signal and the converted analogue signal is reconverted into a digital signal using a second sampling frequency, or, a conversion is effected at the digital level. However, the conversion of the sampling frequency via an intermediate analogue stage usually leads to a loss in quality and requires expensive D/A and A/D convertors as well as requiring the matching process which is necessary in connection with analogue circuits. One can assume in particular, that the intermediately connected analogue stage and the multiple usage of filters will nullify the essential advantages of digital signal processing, c.f.. KRIEG, Bernhard: Praxis der digitalen Audiotechnik. In: Franzis-Arbeitsbuch 1989, p 33.
Furthermore, methods for the digital conversion of sampling frequencies are known but these can only be realised at relatively high expense and by using a multiplicity of circuits if the ratio between the original frequency and the target frequency is not a whole number. In these cases, a very high intermediate conversion frequency is needed. The realisation of the required filters by calculation of the filter co-efficients represents an especially high outlay since the filters use multiple stages and the number of sets of co-efficients that are necessary grows in proportion to the intermediate frequency.
Furthermore, these methods do not allow different target frequencies to be selected using a single circuit, cf LAGADEC, R.:Digital Sampling Frequency Conversion in Digital Audio, AES Premiere Conference Rey, New York, June 1982.
Consequently, the object of the invention is to provide a method and an arrangement for the digital conversion of sampling frequencies which makes it possible to use a lower intermediate conversion frequency as well as the conversion of different sampling rate ratios for universal application at low cost.